The present invention relates to data communications equipment, e.g., modems. In particular, this invention relates to the transmission of both voice and data signals over the same communications facility.
The co-pending, commonly assigned, U.S. Patent application of Bremer et al. entitled "Simultaneous Analog and Digital Communication," Ser. No. 08/076505, filed on Jun. 14, 1993, describes a simultaneous voice and data (SVD) modem in which a voice signal is added to a data signal for transmission over a communications channel to a receiving modem.
In this simultaneous analog and digital communication system, the data signal to be transmitted is represented by a sequence of data symbols, where each data symbol is associated with a particular N-dimensional signal point value taken from a signal space. Similarly, the analog signal, which is represented by a voice signal, is processed so that it is mapped into the N-dimensional signal space to provide a voice signal point. This voice signal point defines the magnitude and angle of a voice signal vector about the origin of the signal space. The data symbol and the voice signal vector are then added together to select a resultant N-dimensional signal point, which is then transmitted to a far-end modem.
Upon reception of the transmitted N-dimensional signal point, the receiver of the far-end modem detects the embedded data symbol and subtracts the data symbol from the received N-dimensional signal point to yield the voice signal vector. This voice signal vector is then used to recreate the voice signal.
As a result, separate full duplex audio and data channels are maintained within a single Public Switched Telephone Network (PSTN) circuit via the division of the data constellation into audio regions as opposed to discrete data points. The region in which a symbol is transmitted during a given baud time determines the data being sent for that symbol, while the location within the region determines the audio signal being sent during that time period.
In such a system, it is desirable to process the audio signal to increase its immunity to noise and other impairments generated in the PSTN channel. One of the forms of processing available is to reduce the amount of redundancy in the transmitted signal by means of linear prediction--that is, to generate an estimate of the current sample as a linear combination of past samples and then subtract this estimate from the actual current sample. The remainder, or residual, is then transmitted in place of the original signal to a receiver. In addition, information on how to form an estimate is also transmitted. The receiver uses the latter information to regenerate the estimate of the signal, which is then added to the received residual to form a reconstituted original signal.
Conventional linear predictors of speech signals are typically of 8th, 10th, or higher order. The order refers to the number of past samples used to estimate the current sample. In generating the estimate of the current sample, each past sample is multiplied by a "predictor coefficient." The resulting products are then additively combined to provide the estimate of the current sample. The predictor coefficients are themselves generated periodically, based on short-term statistical evaluation of the input samples. Typically, these predictor coefficients are quantized, i.e., restricted to a finite set of values.
Unfortunately, determining the quantized predictor coefficients at any point in time is a complex process--especially within the hardware constraints of a modem. Typically, "ideal", or non-quantized, prediction coefficients are first derived as a function of "normalized" autocorrelation coefficients of each sample. This, in itself, adds significant complexity. Finally, once the ideal prediction coefficients have been generated, the ideal prediction coefficients are quantized via a quantization table.